<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!--
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-- Copyright (c) 2010-2019 Belledonne Communications SARL.
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--
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-- This file is part of Liblinphone.
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--
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-- This program is free software: you can redistribute it and/or modify
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-- it under the terms of the GNU General Public License as published by
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-- the Free Software Foundation, either version 3 of the License, or
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-- (at your option) any later version.
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--
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-- This program is distributed in the hope that it will be useful,
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-- but WITHOUT ANY WARRANTY; without even the implied warranty of
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-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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-- GNU General Public License for more details.
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--
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-- You should have received a copy of the GNU General Public License
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-- along with this program. If not, see <http://www.gnu.org/licenses/>.
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-->
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or -->
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<!-- modify it under the terms of the GNU General Public License as -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version. -->
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<!-- -->
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<!-- This program is distributed in the hope that it will be useful, -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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<!-- GNU General Public License for more details. -->
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<!-- -->
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<!-- You should have received a copy of the GNU General Public License -->
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<!-- along with this program; if not, write to the -->
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<!-- Free Software Foundation, Inc., -->
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<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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<!-- -->
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<!-- Sipp default 'uas' scenario. -->
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<!-- -->
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<scenario name="Basic UAS responder">
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<!-- By adding rrs="true" (Record Route Sets), the route sets -->
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<!-- are saved and used for following messages sent. Useful to test -->
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<!-- against stateful SIP proxies/B2BUAs. -->
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<recv request="INVITE" crlf="true">
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</recv>
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<!-- The '[last_*]' keyword is replaced automatically by the -->
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<!-- specified header if it was present in the last message received -->
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<!-- (except if it was a retransmission). If the header was not -->
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<!-- present or if no message has been received, the '[last_*]' -->
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<!-- keyword is discarded, and all bytes until the end of the line -->
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<!-- are also discarded. -->
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<!-- -->
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<!-- If the specified header was present several times in the -->
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<!-- message, all occurences are concatenated (CRLF seperated) -->
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<!-- to be used in place of the '[last_*]' keyword. -->
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<send>
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<![CDATA[
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SIP/2.0 180 Ringing
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Length: 0
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]]>
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</send>
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<send retrans="500">
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:];tag=[call_number]
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[last_Call-ID:]
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[last_CSeq:]
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 96 97 0 8 101 98
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a=rtpmap:96 speex/16000
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a=fmtp:96 vbr=on
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a=rtpmap:97 speex/8000
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a=fmtp:97 vbr=on
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a=rtpmap:101 telephone-event/16000
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a=rtpmap:98 telephone-event/8000
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]]>
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</send>
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<recv request="ACK"
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optional="true"
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rtd="true"
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crlf="true">
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</recv>
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<recv request="BYE">
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</recv>
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<send>
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<![CDATA[
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SIP/2.0 200 OK
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[last_Via:]
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[last_From:]
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[last_To:]
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[last_Call-ID:]
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[last_CSeq:]
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Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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Content-Length: 0
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]]>
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</send>
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<!-- Keep the call open for a while in case the 200 is lost to be -->
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<!-- able to retransmit it if we receive the BYE again. -->
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<pause milliseconds="1000"/>
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<!-- definition of the response time repartition table (unit is ms) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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</scenario>
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